예제 #1
0
파일: gxfenc.c 프로젝트: szatmary/FFmpeg
static int gxf_write_umf_media_audio(AVIOContext *pb, GXFStreamContext *sc)
{
    avio_wl64(pb, av_double2int(1)); /* sound level to begin to */
    avio_wl64(pb, av_double2int(1)); /* sound level to begin to */
    avio_wl32(pb, 0); /* number of fields over which to ramp up sound level */
    avio_wl32(pb, 0); /* number of fields over which to ramp down sound level */
    avio_wl32(pb, 0); /* reserved */
    avio_wl32(pb, 0); /* reserved */
    return 32;
}
예제 #2
0
파일: soxenc.c 프로젝트: Brhett/FFmpeg
static int sox_write_header(AVFormatContext *s)
{
    SoXContext *sox = s->priv_data;
    AVIOContext *pb = s->pb;
    AVCodecContext *enc = s->streams[0]->codec;
    AVDictionaryEntry *comment;
    size_t comment_len = 0, comment_size;

    comment = av_dict_get(s->metadata, "comment", NULL, 0);
    if (comment)
        comment_len = strlen(comment->value);
    comment_size = (comment_len + 7) & ~7;

    sox->header_size = SOX_FIXED_HDR + comment_size;

    if (enc->codec_id == AV_CODEC_ID_PCM_S32LE) {
        ffio_wfourcc(pb, ".SoX");
        avio_wl32(pb, sox->header_size);
        avio_wl64(pb, 0); /* number of samples */
        avio_wl64(pb, av_double2int(enc->sample_rate));
        avio_wl32(pb, enc->channels);
        avio_wl32(pb, comment_size);
    } else if (enc->codec_id == AV_CODEC_ID_PCM_S32BE) {
        ffio_wfourcc(pb, "XoS.");
        avio_wb32(pb, sox->header_size);
        avio_wb64(pb, 0); /* number of samples */
        avio_wb64(pb, av_double2int(enc->sample_rate));
        avio_wb32(pb, enc->channels);
        avio_wb32(pb, comment_size);
    } else {
        av_log(s, AV_LOG_ERROR, "invalid codec; use pcm_s32le or pcm_s32be\n");
        return -1;
    }

    if (comment_len)
        avio_write(pb, comment->value, comment_len);

    for ( ; comment_size > comment_len; comment_len++)
        avio_w8(pb, 0);

    avio_flush(pb);

    return 0;
}
예제 #3
0
static void put_amf_double(AVIOContext *pb, double d)
{
    avio_w8(pb, AMF_DATA_TYPE_NUMBER);
    avio_wb64(pb, av_double2int(d));
}
예제 #4
0
void ff_amf_write_number(uint8_t **dst, double val)
{
    bytestream_put_byte(dst, AMF_DATA_TYPE_NUMBER);
    bytestream_put_be64(dst, av_double2int(val));
}
예제 #5
0
static int caf_write_header(AVFormatContext *s)
{
    AVIOContext *pb = s->pb;
    AVCodecContext *enc = s->streams[0]->codec;
    CAFContext *caf = s->priv_data;
    AVDictionaryEntry *t = NULL;
    unsigned int codec_tag = ff_codec_get_tag(ff_codec_caf_tags, enc->codec_id);
    int64_t chunk_size = 0;
    int frame_size = enc->frame_size;

    if (s->nb_streams != 1) {
        av_log(s, AV_LOG_ERROR, "CAF files have exactly one stream\n");
        return AVERROR(EINVAL);
    }

    switch (enc->codec_id) {
    case AV_CODEC_ID_AAC:
        av_log(s, AV_LOG_ERROR, "muxing codec currently unsupported\n");
        return AVERROR_PATCHWELCOME;
    }

    if (!codec_tag) {
        av_log(s, AV_LOG_ERROR, "unsupported codec\n");
        return AVERROR_INVALIDDATA;
    }

    if (!enc->block_align && !pb->seekable) {
        av_log(s, AV_LOG_ERROR, "Muxing variable packet size not supported on non seekable output\n");
        return AVERROR_INVALIDDATA;
    }

    if (enc->codec_id != AV_CODEC_ID_MP3 || frame_size != 576)
        frame_size = samples_per_packet(enc->codec_id, enc->channels, enc->block_align);

    ffio_wfourcc(pb, "caff"); //< mFileType
    avio_wb16(pb, 1);         //< mFileVersion
    avio_wb16(pb, 0);         //< mFileFlags

    ffio_wfourcc(pb, "desc");                         //< Audio Description chunk
    avio_wb64(pb, 32);                                //< mChunkSize
    avio_wb64(pb, av_double2int(enc->sample_rate));   //< mSampleRate
    avio_wl32(pb, codec_tag);                         //< mFormatID
    avio_wb32(pb, codec_flags(enc->codec_id));        //< mFormatFlags
    avio_wb32(pb, enc->block_align);                  //< mBytesPerPacket
    avio_wb32(pb, frame_size);                        //< mFramesPerPacket
    avio_wb32(pb, enc->channels);                     //< mChannelsPerFrame
    avio_wb32(pb, av_get_bits_per_sample(enc->codec_id)); //< mBitsPerChannel

    if (enc->channel_layout) {
        ffio_wfourcc(pb, "chan");
        avio_wb64(pb, 12);
        ff_mov_write_chan(pb, enc->channel_layout);
    }

    if (enc->codec_id == AV_CODEC_ID_ALAC) {
        ffio_wfourcc(pb, "kuki");
        avio_wb64(pb, 12 + enc->extradata_size);
        avio_write(pb, "\0\0\0\14frmaalac", 12);
        avio_write(pb, enc->extradata, enc->extradata_size);
    } else if (enc->codec_id == AV_CODEC_ID_AMR_NB) {
        ffio_wfourcc(pb, "kuki");
        avio_wb64(pb, 29);
        avio_write(pb, "\0\0\0\14frmasamr", 12);
        avio_wb32(pb, 0x11); /* size */
        avio_write(pb, "samrFFMP", 8);
        avio_w8(pb, 0); /* decoder version */

        avio_wb16(pb, 0x81FF); /* Mode set (all modes for AMR_NB) */
        avio_w8(pb, 0x00); /* Mode change period (no restriction) */
        avio_w8(pb, 0x01); /* Frames per sample */
    } else if (enc->codec_id == AV_CODEC_ID_QDM2) {
        ffio_wfourcc(pb, "kuki");
        avio_wb64(pb, enc->extradata_size);
        avio_write(pb, enc->extradata, enc->extradata_size);
    }

    if (av_dict_count(s->metadata)) {
        ffio_wfourcc(pb, "info"); //< Information chunk
        while ((t = av_dict_get(s->metadata, "", t, AV_DICT_IGNORE_SUFFIX))) {
            chunk_size += strlen(t->key) + strlen(t->value) + 2;
        }
        avio_wb64(pb, chunk_size + 4);
        avio_wb32(pb, av_dict_count(s->metadata));
        t = NULL;
        while ((t = av_dict_get(s->metadata, "", t, AV_DICT_IGNORE_SUFFIX))) {
            avio_put_str(pb, t->key);
            avio_put_str(pb, t->value);
        }
    }

    ffio_wfourcc(pb, "data"); //< Audio Data chunk
    caf->data = avio_tell(pb);
    avio_wb64(pb, -1);        //< mChunkSize
    avio_wb32(pb, 0);         //< mEditCount

    avio_flush(pb);
    return 0;
}
예제 #6
0
파일: ffmenc.c 프로젝트: Brhett/FFmpeg
static int ffm_write_header(AVFormatContext *s)
{
    FFMContext *ffm = s->priv_data;
    AVStream *st;
    AVIOContext *pb = s->pb;
    AVCodecContext *codec;
    int bit_rate, i;

    ffm->packet_size = FFM_PACKET_SIZE;

    /* header */
    avio_wl32(pb, MKTAG('F', 'F', 'M', '1'));
    avio_wb32(pb, ffm->packet_size);
    avio_wb64(pb, 0); /* current write position */

    avio_wb32(pb, s->nb_streams);
    bit_rate = 0;
    for(i=0;i<s->nb_streams;i++) {
        st = s->streams[i];
        bit_rate += st->codec->bit_rate;
    }
    avio_wb32(pb, bit_rate);

    /* list of streams */
    for(i=0;i<s->nb_streams;i++) {
        st = s->streams[i];
        avpriv_set_pts_info(st, 64, 1, 1000000);

        codec = st->codec;
        /* generic info */
        avio_wb32(pb, codec->codec_id);
        avio_w8(pb, codec->codec_type);
        avio_wb32(pb, codec->bit_rate);
        avio_wb32(pb, codec->flags);
        avio_wb32(pb, codec->flags2);
        avio_wb32(pb, codec->debug);
        /* specific info */
        switch(codec->codec_type) {
        case AVMEDIA_TYPE_VIDEO:
            avio_wb32(pb, codec->time_base.num);
            avio_wb32(pb, codec->time_base.den);
            avio_wb16(pb, codec->width);
            avio_wb16(pb, codec->height);
            avio_wb16(pb, codec->gop_size);
            avio_wb32(pb, codec->pix_fmt);
            avio_w8(pb, codec->qmin);
            avio_w8(pb, codec->qmax);
            avio_w8(pb, codec->max_qdiff);
            avio_wb16(pb, (int) (codec->qcompress * 10000.0));
            avio_wb16(pb, (int) (codec->qblur * 10000.0));
            avio_wb32(pb, codec->bit_rate_tolerance);
            avio_put_str(pb, codec->rc_eq ? codec->rc_eq : "tex^qComp");
            avio_wb32(pb, codec->rc_max_rate);
            avio_wb32(pb, codec->rc_min_rate);
            avio_wb32(pb, codec->rc_buffer_size);
            avio_wb64(pb, av_double2int(codec->i_quant_factor));
            avio_wb64(pb, av_double2int(codec->b_quant_factor));
            avio_wb64(pb, av_double2int(codec->i_quant_offset));
            avio_wb64(pb, av_double2int(codec->b_quant_offset));
            avio_wb32(pb, codec->dct_algo);
            avio_wb32(pb, codec->strict_std_compliance);
            avio_wb32(pb, codec->max_b_frames);
            avio_wb32(pb, codec->mpeg_quant);
            avio_wb32(pb, codec->intra_dc_precision);
            avio_wb32(pb, codec->me_method);
            avio_wb32(pb, codec->mb_decision);
            avio_wb32(pb, codec->nsse_weight);
            avio_wb32(pb, codec->frame_skip_cmp);
            avio_wb64(pb, av_double2int(codec->rc_buffer_aggressivity));
            avio_wb32(pb, codec->codec_tag);
            avio_w8(pb, codec->thread_count);
            avio_wb32(pb, codec->coder_type);
            avio_wb32(pb, codec->me_cmp);
            avio_wb32(pb, codec->me_subpel_quality);
            avio_wb32(pb, codec->me_range);
            avio_wb32(pb, codec->keyint_min);
            avio_wb32(pb, codec->scenechange_threshold);
            avio_wb32(pb, codec->b_frame_strategy);
            avio_wb64(pb, av_double2int(codec->qcompress));
            avio_wb64(pb, av_double2int(codec->qblur));
            avio_wb32(pb, codec->max_qdiff);
            avio_wb32(pb, codec->refs);
            break;
        case AVMEDIA_TYPE_AUDIO:
            avio_wb32(pb, codec->sample_rate);
            avio_wl16(pb, codec->channels);
            avio_wl16(pb, codec->frame_size);
            avio_wl16(pb, codec->sample_fmt);
            break;
        default:
            return -1;
        }
        if (codec->flags & CODEC_FLAG_GLOBAL_HEADER) {
            avio_wb32(pb, codec->extradata_size);
            avio_write(pb, codec->extradata, codec->extradata_size);
        }
    }

    /* flush until end of block reached */
    while ((avio_tell(pb) % ffm->packet_size) != 0)
        avio_w8(pb, 0);

    avio_flush(pb);

    /* init packet mux */
    ffm->packet_ptr = ffm->packet;
    ffm->packet_end = ffm->packet + ffm->packet_size - FFM_HEADER_SIZE;
    av_assert0(ffm->packet_end >= ffm->packet);
    ffm->frame_offset = 0;
    ffm->dts = 0;
    ffm->first_packet = 1;

    return 0;
}
예제 #7
0
파일: aiffenc.c 프로젝트: 0day-ci/FFmpeg
static int aiff_write_header(AVFormatContext *s)
{
    AIFFOutputContext *aiff = s->priv_data;
    AVIOContext *pb = s->pb;
    AVCodecParameters *par;
    uint64_t sample_rate;
    int i, aifc = 0;

    aiff->audio_stream_idx = -1;
    for (i = 0; i < s->nb_streams; i++) {
        AVStream *st = s->streams[i];
        if (aiff->audio_stream_idx < 0 && st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
            aiff->audio_stream_idx = i;
        } else if (st->codecpar->codec_type != AVMEDIA_TYPE_VIDEO) {
            av_log(s, AV_LOG_ERROR, "AIFF allows only one audio stream and a picture.\n");
            return AVERROR(EINVAL);
        }
    }
    if (aiff->audio_stream_idx < 0) {
        av_log(s, AV_LOG_ERROR, "No audio stream present.\n");
        return AVERROR(EINVAL);
    }

    par = s->streams[aiff->audio_stream_idx]->codecpar;

    /* First verify if format is ok */
    if (!par->codec_tag)
        return -1;
    if (par->codec_tag != MKTAG('N','O','N','E'))
        aifc = 1;

    /* FORM AIFF header */
    ffio_wfourcc(pb, "FORM");
    aiff->form = avio_tell(pb);
    avio_wb32(pb, 0);                    /* file length */
    ffio_wfourcc(pb, aifc ? "AIFC" : "AIFF");

    if (aifc) { // compressed audio
        if (!par->block_align) {
            av_log(s, AV_LOG_ERROR, "block align not set\n");
            return -1;
        }
        /* Version chunk */
        ffio_wfourcc(pb, "FVER");
        avio_wb32(pb, 4);
        avio_wb32(pb, 0xA2805140);
    }

    if (par->channels > 2 && par->channel_layout) {
        ffio_wfourcc(pb, "CHAN");
        avio_wb32(pb, 12);
        ff_mov_write_chan(pb, par->channel_layout);
    }

    put_meta(s, "title",     MKTAG('N', 'A', 'M', 'E'));
    put_meta(s, "author",    MKTAG('A', 'U', 'T', 'H'));
    put_meta(s, "copyright", MKTAG('(', 'c', ')', ' '));
    put_meta(s, "comment",   MKTAG('A', 'N', 'N', 'O'));

    /* Common chunk */
    ffio_wfourcc(pb, "COMM");
    avio_wb32(pb, aifc ? 24 : 18); /* size */
    avio_wb16(pb, par->channels);  /* Number of channels */

    aiff->frames = avio_tell(pb);
    avio_wb32(pb, 0);              /* Number of frames */

    if (!par->bits_per_coded_sample)
        par->bits_per_coded_sample = av_get_bits_per_sample(par->codec_id);
    if (!par->bits_per_coded_sample) {
        av_log(s, AV_LOG_ERROR, "could not compute bits per sample\n");
        return -1;
    }
    if (!par->block_align)
        par->block_align = (par->bits_per_coded_sample * par->channels) >> 3;

    avio_wb16(pb, par->bits_per_coded_sample); /* Sample size */

    sample_rate = av_double2int(par->sample_rate);
    avio_wb16(pb, (sample_rate >> 52) + (16383 - 1023));
    avio_wb64(pb, UINT64_C(1) << 63 | sample_rate << 11);

    if (aifc) {
        avio_wl32(pb, par->codec_tag);
        avio_wb16(pb, 0);
    }

    if (par->codec_tag == MKTAG('Q','D','M','2') && par->extradata_size) {
        ffio_wfourcc(pb, "wave");
        avio_wb32(pb, par->extradata_size);
        avio_write(pb, par->extradata, par->extradata_size);
    }

    /* Sound data chunk */
    ffio_wfourcc(pb, "SSND");
    aiff->ssnd = avio_tell(pb);         /* Sound chunk size */
    avio_wb32(pb, 0);                    /* Sound samples data size */
    avio_wb32(pb, 0);                    /* Data offset */
    avio_wb32(pb, 0);                    /* Block-size (block align) */

    avpriv_set_pts_info(s->streams[aiff->audio_stream_idx], 64, 1,
                        s->streams[aiff->audio_stream_idx]->codecpar->sample_rate);

    /* Data is starting here */
    avio_flush(pb);

    return 0;
}
예제 #8
0
파일: cafenc.c 프로젝트: Brhett/FFmpeg
static int caf_write_header(AVFormatContext *s)
{
    AVIOContext *pb = s->pb;
    AVCodecContext *enc = s->streams[0]->codec;
    CAFContext *caf = s->priv_data;
    unsigned int codec_tag = ff_codec_get_tag(ff_codec_caf_tags, enc->codec_id);

    switch (enc->codec_id) {
    case AV_CODEC_ID_AAC:
    case AV_CODEC_ID_AC3:
        av_log(s, AV_LOG_ERROR, "muxing codec currently unsupported\n");
        return AVERROR_PATCHWELCOME;
    }

    switch (enc->codec_id) {
    case AV_CODEC_ID_PCM_S8:
    case AV_CODEC_ID_PCM_S16LE:
    case AV_CODEC_ID_PCM_S16BE:
    case AV_CODEC_ID_PCM_S24LE:
    case AV_CODEC_ID_PCM_S24BE:
    case AV_CODEC_ID_PCM_S32LE:
    case AV_CODEC_ID_PCM_S32BE:
    case AV_CODEC_ID_PCM_F32LE:
    case AV_CODEC_ID_PCM_F32BE:
    case AV_CODEC_ID_PCM_F64LE:
    case AV_CODEC_ID_PCM_F64BE:
    case AV_CODEC_ID_PCM_ALAW:
    case AV_CODEC_ID_PCM_MULAW:
        codec_tag = MKTAG('l','p','c','m');
    }

    if (!codec_tag) {
        av_log(s, AV_LOG_ERROR, "unsupported codec\n");
        return AVERROR_INVALIDDATA;
    }

    if (!enc->block_align && !pb->seekable) {
        av_log(s, AV_LOG_ERROR, "Muxing variable packet size not supported on non seekable output\n");
        return AVERROR_INVALIDDATA;
    }

    ffio_wfourcc(pb, "caff"); //< mFileType
    avio_wb16(pb, 1);         //< mFileVersion
    avio_wb16(pb, 0);         //< mFileFlags

    ffio_wfourcc(pb, "desc");                         //< Audio Description chunk
    avio_wb64(pb, 32);                                //< mChunkSize
    avio_wb64(pb, av_double2int(enc->sample_rate));   //< mSampleRate
    avio_wl32(pb, codec_tag);                         //< mFormatID
    avio_wb32(pb, codec_flags(enc->codec_id));        //< mFormatFlags
    avio_wb32(pb, enc->block_align);                  //< mBytesPerPacket
    avio_wb32(pb, samples_per_packet(enc->codec_id, enc->channels)); //< mFramesPerPacket
    avio_wb32(pb, enc->channels);                     //< mChannelsPerFrame
    avio_wb32(pb, av_get_bits_per_sample(enc->codec_id)); //< mBitsPerChannel

    if (enc->channel_layout) {
        ffio_wfourcc(pb, "chan");
        avio_wb64(pb, 12);
        ff_mov_write_chan(pb, enc->channel_layout);
    }

    if (enc->codec_id == AV_CODEC_ID_ALAC) {
        ffio_wfourcc(pb, "kuki");
        avio_wb64(pb, 12 + enc->extradata_size);
        avio_write(pb, "\0\0\0\14frmaalac", 12);
        avio_write(pb, enc->extradata, enc->extradata_size);
    } else if (enc->codec_id == AV_CODEC_ID_AMR_NB) {
        ffio_wfourcc(pb, "kuki");
        avio_wb64(pb, 29);
        avio_write(pb, "\0\0\0\14frmasamr", 12);
        avio_wb32(pb, 0x11); /* size */
        avio_write(pb, "samrFFMP", 8);
        avio_w8(pb, 0); /* decoder version */

        avio_wb16(pb, 0x81FF); /* Mode set (all modes for AMR_NB) */
        avio_w8(pb, 0x00); /* Mode change period (no restriction) */
        avio_w8(pb, 0x01); /* Frames per sample */
    } else if (enc->codec_id == AV_CODEC_ID_QDM2) {
        ffio_wfourcc(pb, "kuki");
        avio_wb64(pb, enc->extradata_size);
        avio_write(pb, enc->extradata, enc->extradata_size);
    }

    ffio_wfourcc(pb, "data"); //< Audio Data chunk
    caf->data = avio_tell(pb);
    avio_wb64(pb, -1);        //< mChunkSize
    avio_wb32(pb, 0);         //< mEditCount

    avio_flush(pb);
    return 0;
}
예제 #9
0
파일: aiffenc.c 프로젝트: KindDragon/FFmpeg
static int aiff_write_header(AVFormatContext *s)
{
    AIFFOutputContext *aiff = s->priv_data;
    AVIOContext *pb = s->pb;
    AVCodecContext *enc = s->streams[0]->codec;
    uint64_t sample_rate;
    int aifc = 0;

    /* First verify if format is ok */
    if (!enc->codec_tag)
        return -1;
    if (enc->codec_tag != MKTAG('N','O','N','E'))
        aifc = 1;

    /* FORM AIFF header */
    ffio_wfourcc(pb, "FORM");
    aiff->form = avio_tell(pb);
    avio_wb32(pb, 0);                    /* file length */
    ffio_wfourcc(pb, aifc ? "AIFC" : "AIFF");

    if (aifc) { // compressed audio
        enc->bits_per_coded_sample = 16;
        if (!enc->block_align) {
            av_log(s, AV_LOG_ERROR, "block align not set\n");
            return -1;
        }
        /* Version chunk */
        ffio_wfourcc(pb, "FVER");
        avio_wb32(pb, 4);
        avio_wb32(pb, 0xA2805140);
    }

    if (enc->channels > 2 && enc->channel_layout) {
        ffio_wfourcc(pb, "CHAN");
        avio_wb32(pb, 12);
        ff_mov_write_chan(pb, enc->channel_layout);
    }

    /* Common chunk */
    ffio_wfourcc(pb, "COMM");
    avio_wb32(pb, aifc ? 24 : 18); /* size */
    avio_wb16(pb, enc->channels);  /* Number of channels */

    aiff->frames = avio_tell(pb);
    avio_wb32(pb, 0);              /* Number of frames */

    if (!enc->bits_per_coded_sample)
        enc->bits_per_coded_sample = av_get_bits_per_sample(enc->codec_id);
    if (!enc->bits_per_coded_sample) {
        av_log(s, AV_LOG_ERROR, "could not compute bits per sample\n");
        return -1;
    }
    if (!enc->block_align)
        enc->block_align = (enc->bits_per_coded_sample * enc->channels) >> 3;

    avio_wb16(pb, enc->bits_per_coded_sample); /* Sample size */

    sample_rate = av_double2int(enc->sample_rate);
    avio_wb16(pb, (sample_rate >> 52) + (16383 - 1023));
    avio_wb64(pb, UINT64_C(1) << 63 | sample_rate << 11);

    if (aifc) {
        avio_wl32(pb, enc->codec_tag);
        avio_wb16(pb, 0);
    }

    if (enc->codec_tag == MKTAG('Q','D','M','2') && enc->extradata_size) {
        ffio_wfourcc(pb, "wave");
        avio_wb32(pb, enc->extradata_size);
        avio_write(pb, enc->extradata, enc->extradata_size);
    }

    /* Sound data chunk */
    ffio_wfourcc(pb, "SSND");
    aiff->ssnd = avio_tell(pb);         /* Sound chunk size */
    avio_wb32(pb, 0);                    /* Sound samples data size */
    avio_wb32(pb, 0);                    /* Data offset */
    avio_wb32(pb, 0);                    /* Block-size (block align) */

    avpriv_set_pts_info(s->streams[0], 64, 1, s->streams[0]->codec->sample_rate);

    /* Data is starting here */
    avio_flush(pb);

    return 0;
}